void RealtimeAnalyser::convertToByteData(DOMUint8Array* destinationArray) {
// Convert from linear magnitude to unsigned-byte decibels.
unsigned sourceLength = magnitudeBuffer().size();
size_t len = std::min(sourceLength, destinationArray->length());
if (len > 0) {
const double rangeScaleFactor = m_maxDecibels == m_minDecibels
? 1
: 1 / (m_maxDecibels - m_minDecibels);
const double minDecibels = m_minDecibels;
const float* source = magnitudeBuffer().data();
unsigned char* destination = destinationArray->data();
for (unsigned i = 0; i < len; ++i) {
float linearValue = source[i];
double dbMag = AudioUtilities::linearToDecibels(linearValue);
// The range m_minDecibels to m_maxDecibels will be scaled to byte values
// from 0 to UCHAR_MAX.
double scaledValue = UCHAR_MAX * (dbMag - minDecibels) * rangeScaleFactor;
// Clip to valid range.
if (scaledValue < 0)
scaledValue = 0;
if (scaledValue > UCHAR_MAX)
scaledValue = UCHAR_MAX;
destination[i] = static_cast<unsigned char>(scaledValue);
}
}
}
void RealtimeAnalyser::getByteFrequencyData(Uint8Array* destinationArray)
{
ASSERT(isMainThread());
if (!destinationArray)
return;
doFFTAnalysis();
// Convert from linear magnitude to unsigned-byte decibels.
unsigned sourceLength = magnitudeBuffer().size();
size_t len = min(sourceLength, destinationArray->length());
if (len > 0) {
const double RangeScaleFactor = m_maxDecibels == m_minDecibels ? 1.0 : 1.0 / (m_maxDecibels - m_minDecibels);
const float* source = magnitudeBuffer().data();
unsigned char* destination = destinationArray->data();
for (unsigned i = 0; i < len; ++i) {
float linearValue = source[i];
double dbMag = !linearValue ? m_minDecibels : AudioUtilities::linearToDecibels(linearValue);
// The range m_minDecibels to m_maxDecibels will be scaled to byte values from 0 to UCHAR_MAX.
double scaledValue = UCHAR_MAX * (dbMag - m_minDecibels) * RangeScaleFactor;
// Clip to valid range.
if (scaledValue < 0.0)
scaledValue = 0.0;
if (scaledValue > UCHAR_MAX)
scaledValue = UCHAR_MAX;
destination[i] = static_cast<unsigned char>(scaledValue);
}
}
}
void RealtimeAnalyser::doFFTAnalysis() {
DCHECK(isMainThread());
// Unroll the input buffer into a temporary buffer, where we'll apply an
// analysis window followed by an FFT.
size_t fftSize = this->fftSize();
AudioFloatArray temporaryBuffer(fftSize);
float* inputBuffer = m_inputBuffer.data();
float* tempP = temporaryBuffer.data();
// Take the previous fftSize values from the input buffer and copy into the
// temporary buffer.
unsigned writeIndex = m_writeIndex;
if (writeIndex < fftSize) {
memcpy(tempP, inputBuffer + writeIndex - fftSize + InputBufferSize,
sizeof(*tempP) * (fftSize - writeIndex));
memcpy(tempP + fftSize - writeIndex, inputBuffer,
sizeof(*tempP) * writeIndex);
} else {
memcpy(tempP, inputBuffer + writeIndex - fftSize, sizeof(*tempP) * fftSize);
}
// Window the input samples.
applyWindow(tempP, fftSize);
// Do the analysis.
m_analysisFrame->doFFT(tempP);
float* realP = m_analysisFrame->realData();
float* imagP = m_analysisFrame->imagData();
// Blow away the packed nyquist component.
imagP[0] = 0;
// Normalize so than an input sine wave at 0dBfs registers as 0dBfs (undo FFT
// scaling factor).
const double magnitudeScale = 1.0 / fftSize;
// A value of 0 does no averaging with the previous result. Larger values
// produce slower, but smoother changes.
double k = m_smoothingTimeConstant;
k = std::max(0.0, k);
k = std::min(1.0, k);
// Convert the analysis data from complex to magnitude and average with the
// previous result.
float* destination = magnitudeBuffer().data();
size_t n = magnitudeBuffer().size();
for (size_t i = 0; i < n; ++i) {
std::complex<double> c(realP[i], imagP[i]);
double scalarMagnitude = abs(c) * magnitudeScale;
destination[i] = float(k * destination[i] + (1 - k) * scalarMagnitude);
}
}
void RealtimeAnalyser::convertFloatToDb(DOMFloat32Array* destinationArray) {
// Convert from linear magnitude to floating-point decibels.
unsigned sourceLength = magnitudeBuffer().size();
size_t len = std::min(sourceLength, destinationArray->length());
if (len > 0) {
const float* source = magnitudeBuffer().data();
float* destination = destinationArray->data();
for (unsigned i = 0; i < len; ++i) {
float linearValue = source[i];
double dbMag = AudioUtilities::linearToDecibels(linearValue);
destination[i] = float(dbMag);
}
}
}
void RealtimeAnalyser::doFFTAnalysis()
{
ASSERT(isMainThread());
// Unroll the input buffer into a temporary buffer, where we'll apply an analysis window followed by an FFT.
size_t fftSize = this->fftSize();
AudioFloatArray temporaryBuffer(fftSize);
float* inputBuffer = m_inputBuffer.data();
float* tempP = temporaryBuffer.data();
// Take the previous fftSize values from the input buffer and copy into the temporary buffer.
// FIXME : optimize with memcpy().
unsigned writeIndex = m_writeIndex;
for (unsigned i = 0; i < fftSize; ++i)
tempP[i] = inputBuffer[(i + writeIndex - fftSize + InputBufferSize) % InputBufferSize];
// Window the input samples.
applyWindow(tempP, fftSize);
// Do the analysis.
m_analysisFrame->doFFT(tempP);
float* realP = m_analysisFrame->realData();
float* imagP = m_analysisFrame->imagData();
// Blow away the packed nyquist component.
imagP[0] = 0.0f;
// Normalize so than an input sine wave at 0dBfs registers as 0dBfs (undo FFT scaling factor).
const double MagnitudeScale = 1.0 / DefaultFFTSize;
// A value of 0 does no averaging with the previous result. Larger values produce slower, but smoother changes.
double k = m_smoothingTimeConstant;
k = max(0.0, k);
k = min(1.0, k);
// Convert the analysis data from complex to magnitude and average with the previous result.
float* destination = magnitudeBuffer().data();
size_t n = magnitudeBuffer().size();
for (size_t i = 0; i < n; ++i) {
Complex c(realP[i], imagP[i]);
double scalarMagnitude = abs(c) * MagnitudeScale;
destination[i] = float(k * destination[i] + (1.0 - k) * scalarMagnitude);
}
}
void RealtimeAnalyser::getFloatFrequencyData(DOMFloat32Array* destinationArray)
{
ASSERT(isMainThread());
ASSERT(destinationArray);
doFFTAnalysis();
// Convert from linear magnitude to floating-point decibels.
const double minDecibels = m_minDecibels;
unsigned sourceLength = magnitudeBuffer().size();
size_t len = std::min(sourceLength, destinationArray->length());
if (len > 0) {
const float* source = magnitudeBuffer().data();
float* destination = destinationArray->data();
for (unsigned i = 0; i < len; ++i) {
float linearValue = source[i];
double dbMag = !linearValue ? minDecibels : AudioUtilities::linearToDecibels(linearValue);
destination[i] = float(dbMag);
}
}
}
